[Asterisk-Users] Can't make outside call with SIP softphone

[David H]

Hi,

I am can make local extension to and from SIP X-Lite
softphone, but I can't dial out using X-Lite but local
analog works just fine. Here are my conf files any
idea? 
Thanks,
David

my sip.conf
[general]
bindport=5060                   ; UDP Port to bind to
(SIP standard port is 5060)
bindaddr=0.0.0.0                ; IP address to bind
to (0.0.0.0 binds to all)
allow=all

[3000]
type=friend
allow=all
username=3000
secret=my_passwd
host=dynamic
context=sip
dtmfmode=rfc2833

my extension.conf

[globals]
davidHand=>Zap/1
davidVoicemail=>1000 at microfortune
johnHand=>Zap/2
johnVoicemail=>2000 at microfortune
davidout=>Zap/3
johnout=>Zap/4

[internal]
exten => 1000,1,Dial(${davidHand},10,r)
exten => 1000,n,Voicemail(u${davidVoicemail})
exten =>
1000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => 1000,n,Wait(1)
exten => 1000,n,Hangup()
exten => 1000,102,Voicemail(b${davidVoicemail})
exten => 1000,103,Hangup()

exten => 2000,1,Dial(${johnHand},10,r)
exten => 2000,n,Voicemail(u${johnVoicemail})
exten =>
2000,n,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => 2000,n,Wait(1)
exten => 2000,n,Hangup()
exten => 2000,102,Voicemail(b${johnVoicemail})
exten => 2000,103,Hangup()

exten => 3000,1,Dial(SIP/3000,20,tr)
exten => 3000,n, Bye()

exten =>
i,1,Playback(/var/lib/asterisk/sounds/invalid)
exten => i,2,Goto(incoming,s,2)

exten =>
t,1,Playback(/var/lib/asterisk/sounds/vm-goodbye)
exten => t,2,Hangup()

[outgoing]
ignorepat => 9
exten => 9,1,Dial(Zap/3)
exten => 9,n,Congestion()
exten => 9,n,Hangup()

[voicemail]
exten => 2828,1,VoiceMailMain()
exten => 2828,n,Hangup()


[incoming]
exten => s,1,Answer()
exten =>
s,2,Background(/var/lib/asterisk/sounds/vm-enter-num-to-call)
include => internal


[sip]
include => internal

[default]
include => internal
include => outgoing
include => sip


	
		
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